Scribd
Upload
65 views8 pages

Chapter 2

The document discusses the sampling theorem and aliasing. It begins by stating the sampling theorem - that a signal must be sampled at twice its highest frequency to perfectly recover the original signal. Sampling lower results in aliasing, where a frequency takes on a false identity. It then provides an intuitive example using a spoked wheel to illustrate aliasing. The sampling theorem is then mathematically proven. It emphasizes that real signals contain a range of frequencies, so anti-aliasing filters are used before sampling to limit the signal's bandwidth and prevent aliasing. Finally, it discusses practical considerations in implementing sampled systems, such as the need for anti-aliasing filters and sample-and-hold circuits.

Uploaded by

Aldon Jimenez
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
65 views8 pages

Chapter 2

The document discusses the sampling theorem and aliasing. It begins by stating the sampling theorem - that a signal must be sampled at twice its highest frequency to perfectly recover the original signal. Sampling lower results in aliasing, where a frequency takes on a false identity. It then provides an intuitive example using a spoked wheel to illustrate aliasing. The sampling theorem is then mathematically proven. It emphasizes that real signals contain a range of frequencies, so anti-aliasing filters are used before sampling to limit the signal's bandwidth and prevent aliasing. Finally, it discusses practical considerations in implementing sampled systems, such as the need for anti-aliasing filters and sample-and-hold circuits.

Uploaded by

Aldon Jimenez
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 8

DIGITAL SIGNAL PROCESSING

Sampling Theorem and


CHAPTER 2
Aliasing

Objectives
After completing this chapter, you will be able to:

• To Gain an Appreciation of the Various Types of


Signals and Systems
• Analyze The Various Types of Systems
• Learn the Skills and Tools needed to Perform These
Analyses.
• Understand How Computers process signals and
systems

An Intuitive Development

The sampling theorem by C.E. Shannon in 1949 places restrictions on the


frequency content of the time function signal, f(t), and can be simply stated as follows: In
order to recover the signal function f(t) exactly, it is necessary to sample f(t) at a rate
greater than twice its highest frequency component. Practically speaking for example, to
sample an analog signal having a maximum frequency of 2Kc requires sampling at
greater than 4Kc to preserve and recover the waveform exactly. The consequences of
sampling a signal at a rate below its highest frequency component results in a
phenomenon known as aliasing. This concept results in a frequency mistakenly taking on
the identity of an entirely different frequency when recovered. In an attempt to clarify this,
envision the ideal sampler of Figure 1(a), with a sample period of T shown in (b), sampling
the waveform f(t) as pictured in (c). The sampled data points of f’(t) are shown in (d) and
can be defined as the sample set of the continuous function f(t). Note in Figure 1(e) that
another frequency component, a’(t), can be found that has the same sample set of data
points as f’(t) in (d). Because of this it is difficult to determine which frequency a’(t), is truly
being observed. This effect is similar to that observed in western movies when watching
the spoked wheels of a rapidly moving stagecoach rotate backwards at a slow rate. The

8
DIGITAL SIGNAL PROCESSING

effect is a result of each individual frame of film resembling a discrete strobed sampling
operation flashing at a rate slightly faster than that of the rotating wheel. Each observed
sample point or frame catches the spoked wheel slightly displaced from its previous
position giving the effective appearance of a wheel rotating backwards. Again, aliasing is
evidenced and in this example it becomes difficult to determine which is the true rotational
frequency being observed.

9
DIGITAL SIGNAL PROCESSING

On the surface it is easily said that anti-aliasing designs can be achieved by


sampling at a rate greater than twice the maximum frequency found within the signal to
be sampled. In the real world, however, most signals contain the entire spectrum of
frequency components; from the desired to those present in white noise. To recover such
information accurately the system would require an unrealizably high sample rate. This
difficulty can be easily overcome by preconditioning the input signal, the means of which
would be a band-limiting or frequency filtering function performed prior to the sample data
input. The prefilter, typically called anti-aliasing filter guarantees, for example in the low
pass filter case, that the sampled data system receives analog signals having a spectral
content no greater than those frequencies allowed by the filter. As illustrated in Figure 2,
it thus becomes a simple matter to sample at greater than twice the maximum frequency
content of a given signal. A parallel analog of band-limiting can be made to the world of
perception when considering the spectrum of white light. It can be realized that the study
of violet light wavelengths generated from a white light source would be vastly simplified
if initial band-limiting were performed through the use of a prism or white light filter.

The Sampling Theorem

To solidify some of the intuitive thoughts presented in the previous section, the
sampling theorem will be presented applying the rigor of mathematics supported by an
illustrative proof. This should hopefully leave the reader with a comfortable understanding
of the sampling theorem.

10
DIGITAL SIGNAL PROCESSING

11
DIGITAL SIGNAL PROCESSING

12
DIGITAL SIGNAL PROCESSING

Eq (15) is equivalent to eq (2) as is illustrated in Figure 4e and Figure 3a


respectively. As observed in Figures 3 and 4, each step of the sampling theorem proof
was also illustrated with its Fourier transform pair. This was done to present alternate
illustrative proofs. Recalling the convolution2 theorem, the convolution of F(0), Figure 3b,
with a set of equidistant impulses, Figure 3d, yields the same periodic frequency function
Fp(0), Figure 3f, as did the Fourier transform of fn, Figure 3e, the product of f(t), Figure
3a, and its equidistant sample impulses, Figure 3c. In the same light the original time
function f(t), Figure 4e, could have been recovered from its sampled waveform by
convolving fn, Figure 4a, with h(t), Figure 4c, rather than multiplying Fp(0), Figure 4b, by
the rectangular function H(0), Figure 4d, to get F(0), Figure 4f, and finally inverse
transforming to achieve f(t), Figure 4e, as done in the mathematic proof.

The Sampling Theorem and Its Hardware Implications

Though there are numerous sophisticated techniques of implementation, it is


appropriate to re-emphasize that the intent of this article is to give the first time user a
basic and fundamental approach toward the design of a sampled-data system. The
method with which to achieve this goal will be to introduce a few of the common perils
encountered when implementing such a system. We begin by considering the generalized
13
DIGITAL SIGNAL PROCESSING

block diagram of Figure 7. As shown inFigure 7, prior to any signal processing


manipulation the analog input signal must be preconditioned to prevent aliasing and
thereafter digitized to logic signals usable by the logic function block. The antialiasing and
digitizing functions are performed by an input filter and analog-to-digital converter
respectively. Once digitized the signal can then be altered or processed and upon
completion, reconstructed back to a continuous analog signal via a digital-to-analog
converter followed by a smoothing filter.

To this point no mention has been made concerning the sample and hold circuit
block depicted in Figure 7. In general the analog-to-digital converter can operate as a
stand-alone unit. In many high speed operations however, the converter speed is
insufficient and thus requires the assistance of a sample and hold circuit.

The Antialiasing Input Filter

As indicated earlier in the text, the antialiasing filter should band-limit the input signal’s
spectrum to frequencies no greater than the Nyquist frequency. In the real world however,
filters are non-ideal and have typical attenuation or bandlimiting and phase characteristics
as shown in Figure 8. It must also be realized that true band-limiting of a specific
frequency spectrum is not possible. In the sample data system band-limiting is achieved
by attenuating those frequencies greater than the Nyquist frequency to a level

14
DIGITAL SIGNAL PROCESSING

undetectable or invisible to the system analog-to-digital (A/D) converter. This level would
typically be less than the rms quantization noise level defined by the specific converter
being used.

As an example of how an antialiasing filter would be applied, assume a sample


data system having within it an 8-bit A/D converter. Eight bits translates to 2ne28e256
levels of resolution. If a 2.56 volt reference were used each quantization level, q, would
represent the equivalent of 2.56 volts/ 256e10 millivolts. Realizing this the antialiasing
filter would be designed such that frequencies in the stopband were attenuated to less
than the rms quantization noise level of q/203 and thus appearing invisible to the system.

Reference:
▪ https://web.ece.ucsb.edu/Faculty/rodwell/Classes/ece2c/resources/an-
236.pdf
Video Tutorial Link:
Sampling Theorem
▪ https://www.youtube.com/watch?v=iQaFDpiNOlA
Sampling and Aliasing
▪ https://www.youtube.com/watch?v=yWqrx08UeUs

15

You might also like